1. Field of the Invention
The present invention relates generally to a method and arrangement of eliminating delayed echoes in a digital telecommunications system, and more specifically to such an arrangement and method by which delayed "network echoes" can effectively be canceled without incurring undesirable increase in the size of an adaptive digital filter as compared with the size of the counterpart provided in an analog telecommunications system. The term "network echoes" implies the echoes which are mainly caused by impedance mismatches in distant four-wire to two-wire hybrid(s) of public switched telephone networks.
2. Description of the Prior Art
Long distance telephone communications systems have been constantly plagued by the so called "echo" phenomenon. A known approach to solving this problem is the use of an echo canceler which by means of an adaptive digital filter makes an estimate of the transfer function of the echo path and uses that information to subtract the echo in the return path.
Before turning to the present invention it is deemed advantageous to describe a known echo canceler for use in a mobile telephone terminal of an analog mobile telephone communications system with reference to FIG. 1.
It should be noted that while the FIG. 1 arrangement is disclosed in connection with a vehicle mounted speakerphone, the present invention is not limited to such applications and can be applied to a wide variety of telecommunication systems.
The arrangement shown in FIG. 1 includes, an antenna 10, a duplexer 12, a front end 14, four amplifiers 16a-16d, two codecs 18, 20, a network echo canceler 22, an acoustic echo canceler 24, a loudspeaker 26 and a microphone 28, all of which are coupled as illustrated. A "codec" is defined as an assembly comprising an encoder and a decoder in the same equipment (CCITT recommendations). Merely by way of example, each of the codecs 18, 20 in this instance, is of a 64 kbit/sec .mu.-law (viz., A-law) PCM (pulse code modulation) type.
The codec 18 includes an analog/digital (A/D) converter 18a and a digital/analog (D/A) converter 18b. Similarly, the other codec 20 is provided with a D/A converter 20a and an A/D converter 20b. Further, the network echo canceler 22 includes an adaptive digital filter (ADF) 22a and an adder 22b, while the acoustic echo canceler 24 includes an ADF 24a and an adder 24b.
The front end (viz., radio signal section) 14 is supplied with an incoming radio signal (analog) via the antenna 12 and the duplexer 12. The analog signal outputted from the front end 14, is applied to an amplifier 16a by which the amplitude of the incoming signal is adjusted to a predetermined one. The A/D converter 18a converts the analog output of the amplifier 16a to the corresponding digital signal.
The ADF 22a of the network echo canceler 22 is arranged to receive the output of the acoustic echo canceler 24 and outputs an estimated (or synthetic) echo signal 22c. The network echo canceler 22 operates such as to eliminate the network echo by rendering an error signal Se zero. As is well known in the art, the network echo canceler 22 updates tap-coefficients of the ADF 22a using the learning algorithms which are demonstrated by the following equation (1). EQU H.sub.n+1 =H.sub.n +.alpha.*(Xn/XnXn.sup.T)*Se.sub.n ( 1)
Where: H.sub.n is a row vector of tap-coefficient matrix at a time "n", Xn a row vector of a matrix of the digitized incoming signal at a time n, Se.sub.n an error signal at a time n, .alpha. a convergence coefficient (0&lt;.alpha.&lt;2), T indicates transposition of vector, and * indicates convolution.
When near-end and distant-end parties talk simultaneously, i.e., during double talk, the network echo canceler 22 is no longer able to correctly update tap-coefficients of the ADF 22a. This is because the distant party's signals mask the network echo. Accordingly, it is a current practice to provide a double talk detector to inhibit the updating of tap-coefficients of the ADF 22a while such is detected.
The aforesaid tap-coefficients updating and the double talk detection, are well known in the art and are not directly concerned with the present invention and, hence, further descriptions thereof will be omitted for simplifying the instant disclosure.
In the case where the tap-coefficients updating is correctly implemented, the distant party's speech signal is derived from the adder 22b during the double talk while effectively canceling the network echoes. The output of the adder 22b is applied to the ADF 24a of the acoustic echo canceler 20 and also is converted into the corresponding analog signal at the D/A converter 20a. The output of the D/A converter 20a is amplified by the amplifier 16b and then drives the loudspeaker 26.
A speech signal issued from the microphone 28 is amplified by the amplifier 16c and then converted into the corresponding digital signal at the A/D converter 20b. The adder 24b of the acoustic echo canceler 24 is arranged to subtract an estimated echo 24c (viz., the output of the ADF 24a) from the output of the A/D converter 20b. The acoustic echo canceler 24 operates in a manner which establishes an error signal Se' and eliminates the acoustic echo which is established via an acoustic coupling between the loudspeaker 26 and the microphone 28. The acoustic echo canceler 24 updates tap-coefficients of the ADF 24a in the same manner as in the network echo canceler 22 using the above mentioned learning algorithms shown by equation (1).
In the event that the tap-coefficients updating of the acoustic echo canceler 24 is correctly implemented, the near-by party's speech signal is derived from the adder 24b while the acoustic echoes are effectively canceled in a manner whereby no subjective interference to the telephone conversation occurs. The output of the adder 24b is applied to the ADF 22a and is also converted into the corresponding analog signal at the D/A converter 18b of the codec 18. The output of the D/A converter 18a is amplified by the amplifier 16d and then transmitted to the distant-end party (not shown) via the front end 14, the duplexer 12 and the antenna 10.
In the above mentioned arrangement (FIG. 1), the network echo is caused by a distant four-wire to two-wire hybrid provided in the vicinity of the distant-end party and is delayed by about 40 ms when returning to the network echo canceler 22. In the event that a sampling clock used in the FIG. 1 arrangement is 8 kHz, the number of samples derived during the 40 ms period is 320. Accordingly, the ADF 22a is required to have 320 taps in order to cancel the delayed network echo (40 ms).
FIG. 2 is a block diagram showing a digital type mobile terminal wherein the codec 18 of FIG. 1 is replaced with a speech codec 30.
The speech codec 30 in this instance takes the form of an LPC (linear predictive coding) codec such as a 11.2 kbit/s VSELPC (vector sum exited LPC) type. The ADF 22a of FIG. 1 should be modified to cancel the network echo whose delay time is prolonged due to the provision of the speech codec 30 and thus, in FIG. 2, a prime (') is added to the reference numerals 22, 22a for differentiating same from the counterparts of FIG. 1.
As is well known in the art, this type of coding offers low data rates ranging from 8-16 kbit/s (for example). This means that the 40 ms delayed network echo is further delayed by about 100 ms at the speech codec 30, and thus the total delay time of the network echo amounts to approximately 140 ms. Accordingly, if the sampling clock used in the FIG. 2 arrangement is 8 kHz (viz., the same as in the FIG. 1 arrangement), the number of samples taken during 140 ms is 1120. Accordingly, the ADF 22a must be provided with 1120 taps in order to cancel the network echo which is delayed by 140 ms.
Further, the network echo canceler 22 requires, for echo cancellation, 4 instruction steps per tap (viz., one instruction step for producing the estimated echo signal and three instruction steps for updating tap coefficients). This means that the network echo canceler 22 requires a computing power of about 36 MIPS (million instructions per second) (viz., 1120.times.4.times.8000=35.84 millions).
Still further, in order to realize the adaptive digital filtering at the ADF 22a, it is necessary to provide a RAM (random access memory) with a memory capacity of 2,240 words for storing the 1120 tap coefficients and 1120 sampled data.
It is understood from the foregoing that the ADF 22a of FIG. 2 is rendered undesirably bulky, complex as compared with the FIG. 1 arrangement.